Why do people listen to music with high sampling frequencies?What is 88k+ providing them?I understand why you might record at a high sampling rate, but why keep that for playback?

Looking through the FAQ, there are threads from 2003 that point out that the sampling frequency and bit-dept work in tandem. So the quantisation error of 16bit at 44.1k has the opportunity to be corrected sooner at a higher sampling rate, so in some ways is like a dithering pattern.

However given noise introduced in the analogue systems required to listen to music, a SNR within a 16bit signal of ~96dB seems pretty good.

So assuming that speakers struggle to produce the sounds that a 192k sampling frequency allow (eg 96kHz) and assuming that 16bits were sufficient when compared to the analogue equipment in the system, what have I missed in these high sampling playback formats?

There is a school of thought that the harsh sound of cds is due to the brickwall filter at around 20kHz and the effects of it's phase/time aberrations. With 96kHz, the filter only needs to cut off by 48kHz so a gentle slope from 20kHz-48kHz is used with much less phase/time aberrations.

1) The vast majority of ADC and DAC chips employ linear phase FIR filters for anti-aliasing and anti-imaging. So there are no phase/time aberrations.

2) While one could use a shallow slope rolling off between 20kHz and 40kHz with 2x sample rate, this is not what the vast majority of ADCs or DACs do. They cut off just as steeply (well, almost) as in the 1x sample case.

QUOTE (icstm @ Feb 20 2012, 14:43)

So use higher sampling frequencies in the capturing and the digitisation of the music, then store for playback at 44.1? surely?

Prior to 'storing for playback at 44.1kHz' one has to downsample, which implies and includes steep anti-alias filtering at 22.05kHz. Sort of back to square 1, not?

Have we conveniently forgotten that CD players commonly employed over sampling in order to aid in reconstruction?

Let's compare and contrast oversampling which is what just about all modern digital<->analog converters do, with upsampling or simply sampling the same analog signal at a far higher rate.

An oversampled converter can be thought of as a black box that is an interface between digital data at a certain data rate, and analog data that for all the world seems to have been sampled at that same sample rate. If we apply the rules of ducks to these gizmos in a standard red book CD player, they walk like they run at 44.1 KHz, they quack like they run at 44.1 KHz, and they look like they run at 44.1 KHz. For all practical purposes they [b]are[/b} 44.1 KHz converters. To a certain degree, what happens inside of them is none of our concern.

Upsampling is a completely different thing. If we think of this process as a black box, it is an interface between stuff that for all the world seems to have been sampled at one sample rate and is now seems to be at some other sample rate. Only by analyzing the data at some deeper level of detail do we get the harsh surprise that first impressions are wrong, and the data actually has properties that are the same or a little worse than it had at the lower sample rate.

Sampling at a higher data rate is again a completely different thing. If we think of this process as a black box, it is an interface between stuff that for all the world seems to have been sampled at a sample rate and is now seems to be like data taken at the same sample rate. However there are no harsh surprises in terms of data. The data really has the properties of data taken at the sample rate that is right there before us.

However, when we are talking about audio signals that are presented with 44.1 KHz sampling and 16 bit resolution, there may still be some harsh surprises. It turns out that due to some inherent limitations of our ears, the data is generally indistinguishable from data taken at even a lower sample rate (ca. 16 Khz), and with less resolution (ca. 13-14 bits).

Hey I didn't make this world, I just try to make reliable observations of it! ;-)